1. Field of the Invention
The present invention relates to an adaptive system in a digital data receiver providing compensation for amplitude and phase distortions introduced by a data transmission channel.
2. Description of the Prior Art
It is a known fact that at high transmission rates a correct restitution, at the output of the data transmission channel, of the signals applied to its input is no longer possible without the provision of a compensation circuit, designated equalizer, which is often constituted by a non-recursive transversal filter, that is to say a circuit capable of correcting the response of a transmission channel on the basis of a finite and weighted sum of partial responses available on the consecutive taps of an impedance network based on delay lines. An equalizer of a conventional type, having N weighting coefficients is shown in FIG. 1 (a description of such an equalizer having seven coefficients is given in the publication IEEE Transactions on Information Theory, Vol. IT-15, No. 4, July 1969, pages 484 to 497). Since the impulse response of the channel is not known and furthermore tends to evolve in time, the equalizer must be adaptive, that is to say it must be capable of adjusting its weighting coefficients to their optimum values at the beginning of the transmission (during the acquisition or training phase of the equalizer) and of following thereafter any variations of the channel during the actual transmission phase. This adaptivity finds expression in that the equalizer generates an error signal which is a function of the difference between the exact form of the transmitted digital data and the form they have to the output of the equalizer, and is arranged so as to reduce this error to a minimum.
In order to provide an efficient use of the receiving system, the training phase must be as short as possible, which means that the method of determining the optimum coefficients of the adaptive equalizer must converge as rapidly as possible. Because of their aptitude in following the temporal variations of the data transmission channel, iterative determination methods are frequently employed, such as the stochastic method. But the convergence speed of this method decreases according as the eigenvalues of the signal autocorrelation matrix A of the output signal of the transmission channel are more dispersed, that is to say according as the amplitude distortion introduced by the channel is more important. If the channel were perfect and its spectrum perfectly flat, the distortion would be zero and all the eigenvalues of A would be equal to 1. In reality, as soon as the channel introduces a significant amplitude distortion (or as soon as intersymbol interference is deliberately created for spectrum shaping purposes), the use of the stochastic gradient method becomes ineffectual.
A satisfactory convergence rate may be obtained by using an iterative equalization method of the self-orthogonalizing type as described in the article by R. D. Gitlin and F. R. Magee "Self-Orthogonalizing Adaptive Equalization Algorithms", published in IEEE Transactions on Communications, Vol. COM-25, No. 7, July 1977, pages 666 to 672, the Kalman filter (applied to the field of equalization by D. Godard, see in this respect reference [16] on page 672 of said article) constituting a special case of this method. However, compared to the stochastic gradient method this novel method is characterized by the fact that the circuits required for its implementation are much more complicated and by the fact that the number of operations to be performed in the course of each iteration is increased considerably.